Timothy Smith
2009-May-16 11:46 UTC
[asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands of dollars to add those to our cisco call manager 4.1 set up. I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), and also a dialpeer to forward on the router to forward calls to my asterisk. It works properly but the problem is there is NO AUDIO! I have tried to change codec but no sucess! Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Kind Regards, Wilson -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: show-run.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/672c1104/attachment.txt -------------- next part -------------- cs-intranet*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 103 172.17.3.249 5060 OK (3 ms) 102 172.17.3.248 5060 OK (3 ms) 101 172.17.10.150 5060 OK (1 ms) 100/100 172.19.4.102 D N 32544 Unmonitored 4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline] ; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP phone -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: show-dialpeer.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/672c1104/attachment-0001.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.conf Type: application/octet-stream Size: 3327 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/672c1104/attachment.obj
On 16 May 2009, at 12:46, Timothy Smith wrote:> <blah> > > Has anyone had the above set up working successfully? Attached are > some confs. > > Thanks a lot for your assistance.Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by now..
David Backeberg
2009-May-16 13:02 UTC
[asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com> wrote:> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), > and also a dialpeer to forward on the router to forward calls to my > asterisk. It works properly but the problem is there is NO AUDIO! I > have tried to change codec but no sucess! > Has anyone had the above set up working successfully?Yes. You have been caught by a not-very-well-documented issue with setting up voice routing on the 3845, and probably other similar Cisco gear. And I'm not sure how you've done your test. This is the closest I've ever seen to a document that explains your problem: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml Did you have a SIP phone on one side of asterisk and a POTS phone on the outside of the 3845? If you did, and you could talk on both at the same time, I think you would discover in fact that you do have some audio, in fact, one-way audio to be precise. But I don't remember for sure, because it's been a while since I've done this to myself. At any rate, your problem is you have dial-peers to get voice packets out from the 3845 to Cisco, but no dial-peers to get the packets from SIP back to a physical circuit on the 3845. Think about this. What should happen to a call inbound from asterisk, to the 3845? Should it go out an E1 to the outside phones world? If so, you need to build a dial-peer that does that. Until you do, you won't be getting two-way audio. you need another rule something like: dial-peer voice 790792888 pots map this back to the proper E1 circuit A secondary problem could also be with the way you're managing your DSPs. I don't know how many physical DSPs you have in your router, but usually it's a GOOD thing to enable DSP farming.
Hi, We have AS5400's set up with asterisk boxes. Initially we had similar issues, but as described, you need to have dial peers to handle both incoming and outgoing peers. Please post your dial peer configs as well as the serial interface configs. I also found that until I add [isdn incoming-voice modem ] I could not get incoming calls on that serial interface to route to my * box. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Saturday, 16 May 2009 10:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 58, Issue 40 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request at lists.digium.com You can reach the person managing the list at asterisk-users-owner at lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 2. Re: Fwd: Asterisk With Cisco Voice Router (Steve Howes) 3. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 4. Re: Fwd: Asterisk With Cisco Voice Router (David Backeberg) 5. Re: meetme dies looking for conf-getconfno (sean darcy) 6. howto set up persistent dynamic meetme (sean darcy) 7. Agent-Login/out in 1.6 (David Anthony O Reilly) 8. Agent-Login/out in 1.6 (David Anthony O Reilly) 9. Re: Agent-Login/out in 1.6 (Stefan Reuter) 10. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 11. Re: Agent-Login/out in 1.6 (Jim Dickenson) 12. Re: howto set up persistent dynamic meetme (Tilghman Lesher) 13. Re: Fwd: Asterisk With Cisco Voice Router (Philipp Kempgen) ---------------------------------------------------------------------- Message: 1 Date: Sat, 16 May 2009 14:46:27 +0300 From: Timothy Smith <timotsmith at gmail.com> Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <416fc8170905160446r5815fd87m67e62506ad9ac031 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands of dollars to add those to our cisco call manager 4.1 set up. I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), and also a dialpeer to forward on the router to forward calls to my asterisk. It works properly but the problem is there is NO AUDIO! I have tried to change codec but no sucess! Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Kind Regards, Wilson -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: show-run.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0002.txt -------------- next part -------------- cs-intranet*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 103 172.17.3.249 5060 OK (3 ms) 102 172.17.3.248 5060 OK (3 ms) 101 172.17.10.150 5060 OK (1 ms) 100/100 172.19.4.102 D N 32544 Unmonitored 4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline] ; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP phone -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: show-dialpeer.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0003.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: sip.conf Type: application/octet-stream Size: 3327 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0001.obj ------------------------------ Message: 2 Date: Sat, 16 May 2009 13:25:40 +0100 From: Steve Howes <steve at geekinter.net> Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <659DC612-4035-4D7E-A73C-77B5A16D607D at geekinter.net> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes On 16 May 2009, at 12:46, Timothy Smith wrote:> <blah> > > Has anyone had the above set up working successfully? Attached are > some confs. > > Thanks a lot for your assistance.Check about the sip.conf 'insecure' option. I have had to use it in the past for similar stuff. I think it was 'insecure=very' but that might be deprecated by now.. ------------------------------ Message: 3 Date: Sat, 16 May 2009 15:46:51 +0300 From: Timothy Smith <timotsmith at gmail.com> Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <416fc8170905160546o6ca0e92fma6361525b244855 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Thanks Steve for this tip. I have insecure=very is not yet deprecated. I have added it but still no good. I personally think the problem could be with the codecs. Any ideas? I have attached some debug info. Regards, Tim On Sat, May 16, 2009 at 3:25 PM, Steve Howes <steve at geekinter.net> wrote:> > On 16 May 2009, at 12:46, Timothy Smith wrote: >> <blah> >> >> Has anyone had the above set up working successfully? Attached are >> some confs. >> >> Thanks a lot for your assistance. > > Check about the sip.conf 'insecure' option. I have had to use it in > the past for similar stuff. I think it was 'insecure=very' but that > might be deprecated by now.. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- VG2# VG2# VG2# VG2# VG2# May 16 12:41:40.237: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref 0x0C73 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98392 Exclusive, Channel 18 Calling Party Number i = 0x2183, '730230199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '790792888' Plan:ISDN, Type:National May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_INCOMING May 16 12:41:40.237: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x33B3, Guid = BCBB464BB098 VG2# May 16 12:41:40.241: fb_get_reject_cause_code: ERROR cause_code NULL May 16 12:41:40.245: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.245: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8C73 Channel ID i = 0xA98392 Exclusive, Channel 18 May 16 12:41:40.353: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:40.353: ISDN Se0/0/1:15 Q931: TX -> ALERTING pd = 8 callref = 0x8C73 VG2# May 16 12:41:46.697: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret May 16 12:41:46.697: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 12:41:46.697: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8C73 May 16 12:41:46.713: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0C73 May 16 12:41:46.713: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_PROGRESS OULKLAVG2# May 16 12:41:46.713: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0C73 Cause i = 0x8490 - Normal call clearing May 16 12:41:52.373: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_DISC May 16 12:41:52.373: %ISDN-6-CONNECT: Interface Serial0/0/1:17 is now connected to 730230199 N/A May 16 12:41:52.373: %ISDN-6-DISCONNECT: Interface Serial0/0/1:17 disconnected from 730230199 , call lasted 5 seconds VG2# May 16 12:41:52.373: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8C73 May 16 12:41:52.385: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0C73 May 16 12:41:52.385: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x33B3 calltype 2 CALL_CLEARED -------------- next part -------------- From: "730230199" <sip:730230199 at 172.19.3.150>;tag=as5f114784 To: <sip:100 at 172.19.4.102:32544;rinstance=e6a140ee2d1dee0f>;tag=ae700477 Contact: <sip:730230199 at 172.19.3.150> Call-ID: 7beff1bd661329c643aa69ec43628adc at 172.19.3.150 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.9 Content-Length: 0 --- -- SIP/100-00820520 answered SIP/172.17.3.248-007fc920 Audio is at 172.19.3.150 port 13312 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 172.17.3.248:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK501204;received=172.17.3.248 From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9 Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:730232888 at 172.19.3.150> Content-Type: application/sdp Content-Length: 261 v=0 o=root 544232458 544232458 IN IP4 172.19.3.150 s=Asterisk PBX 1.6.0.9 c=IN IP4 172.19.3.150 t=0 0 m=audio 13312 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/172.17.3.248-007fc920 and SIP/100-00820520 cs-intranet*CLI> <--- SIP read from UDP://172.17.3.248:62582 ---> ACK sip:730232888 at 172.19.3.150:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK511A76 From: <sip:730230199 at 172.17.3.248>;tag=D8FE7BF8-4CA To: <sip:730232888 at 172.19.3.150>;tag=as0fb38dd9 Date: Sat, 16 May 2009 12:38:27 GMT Call-ID: 4A137712-414D11DE-9606C927-51AF51F3 at 172.17.3.248 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> --- (10 headers 0 lines) --- cs-intranet*CLI> ------------------------------ Message: 4 Date: Sat, 16 May 2009 09:02:46 -0400 From: David Backeberg <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <3de056a30905160602x6002a7a0jc21e67a346338d2d at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com> wrote:> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), > and also a dialpeer to forward on the router to forward calls to my > asterisk. It works properly but the problem is there is NO AUDIO! I > have tried to change codec but no sucess! > Has anyone had the above set up working successfully?Yes. You have been caught by a not-very-well-documented issue with setting up voice routing on the 3845, and probably other similar Cisco gear. And I'm not sure how you've done your test. This is the closest I've ever seen to a document that explains your problem: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0 080147524.shtml Did you have a SIP phone on one side of asterisk and a POTS phone on the outside of the 3845? If you did, and you could talk on both at the same time, I think you would discover in fact that you do have some audio, in fact, one-way audio to be precise. But I don't remember for sure, because it's been a while since I've done this to myself. At any rate, your problem is you have dial-peers to get voice packets out from the 3845 to Cisco, but no dial-peers to get the packets from SIP back to a physical circuit on the 3845. Think about this. What should happen to a call inbound from asterisk, to the 3845? Should it go out an E1 to the outside phones world? If so, you need to build a dial-peer that does that. Until you do, you won't be getting two-way audio. you need another rule something like: dial-peer voice 790792888 pots map this back to the proper E1 circuit A secondary problem could also be with the way you're managing your DSPs. I don't know how many physical DSPs you have in your router, but usually it's a GOOD thing to enable DSP farming. ------------------------------ Message: 5 Date: Sat, 16 May 2009 09:04:36 -0400 From: sean darcy <seandarcy2 at gmail.com> Subject: Re: [asterisk-users] meetme dies looking for conf-getconfno To: asterisk-users at lists.digium.com Message-ID: <gumdl4$o5b$1 at ger.gmane.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed sean darcy wrote:> sean darcy wrote: >> Mark Michelson wrote: >>> sean darcy wrote: >>>> Danny Nicholas wrote: >>>>> You "lost" conf-getconfno.gsm . Asterisk is trying to play thatfile to let>>>>> you pick a conference number to use. It goes in/var/lib/asterisk/sounds.>>>>> Grep for it. >>>>> >>>>> -----Original Message----- >>>>> From: asterisk-users-bounces at lists.digium.com >>>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of seandarcy>>>>> Sent: Friday, May 15, 2009 12:39 PM >>>>> To: asterisk-users at lists.digium.com >>>>> Subject: [asterisk-users] meetme dies looking for conf-getconfno >>>>> >>>>> With 1.6.1, I'm trying to set up a test of meetme for creatingdynamic>>>>> conferences. >>>>> >>>>> cat meetme.conf >>>>> [rooms] >>>>> conf => 600 >>>>> >>>>> extensions.conf: >>>>> >>>>> [meetme] >>>>> exten => 2663,1,MeetMe(,D) >>>>> exten => 2663,n,Hangup() >>>>> >>>>> exten => 2666,1,MeetMe() >>>>> exten => 2666,n,Hangup() >>>>> >>>>> >>>>> What I'm expecting is to dial 2663, get a conference room number (600,>>>>> I suppose since it's the only room ), and set a PIN. >>>>> >>>>> Then I'd dial 2666 enter the conference room number and the PIN,and be>>>>> put in conference. >>>>> >>>>> But here's what happens when I dial 2663: >>>>> >>>>> -- Starting simple switch on 'DAHDI/1-1' >>>>> -- Executing [2663 at internal:1] MeetMe("DAHDI/1-1", ",D") innew stack>>>>> [2009-05-15 13:21:19] WARNING[2061]: file.c:641ast_openstream_full:>>>>> File conf-getconfno does not exist in any format >>>>> [2009-05-15 13:21:19] WARNING[2061]: file.c:924 ast_streamfile:Unable>>>>> to open conf-getconfno (format 0x4 (ulaw)): No such file ordirectory>>>>> == Spawn extension (internal, 2663, 1) exited non-zero on'DAHDI/1-1'>>>>> -- Hungup 'DAHDI/1-1' >>>>> >>>>> >>>>> conf-getconfno does exist: >>>>> >>>>> ls -l /var/lib/asterisk/sounds/en/conf-getconf* >>>>> -rw-r--r--. 1 root root 25211 2009-03-26 14:42 >>>>> /var/lib/asterisk/sounds/en/conf-getconfno.ulaw >>>>> -rw-r--r--. 1 root root 50466 2009-03-26 14:42 >>>>> /var/lib/asterisk/sounds/en/conf-getconfno.wav >>>>> >>>>> Any help appreciated. >>>>> >>>>> sean >>>>> >>>> Will do, but why gsm? Nobody's using gsm, and it looks like it'sseeking>>>> ulaw ( which is installed). >>>> >>>> sean >>>> >>> You may want to try setting languageprefix=yes in asterisk.conf inthe [options]>>> section. If that does not work, then you may wish to try to move thefile up one>>> directory level and see if it plays, then. >>> >>> Mark Michelson >>> >> I did both: >> >> grep language asterisk.conf >> languageprefix = yes ; Use the new sound prefix path syntax >> >> And I copied the files to /var/lib/asterisk/sounds - and then onlygsm>> and then only ulaw. >> >> Restarted. Rebooted. >> >> No luck. >> >> sean >> > > Filed as > > https://issues.asterisk.org/view.php?id=15125 > > sean >Well it works for me now if I enable [directories] in asterisk.conf. sean ------------------------------ Message: 6 Date: Sat, 16 May 2009 09:21:43 -0400 From: sean darcy <seandarcy2 at gmail.com> Subject: [asterisk-users] howto set up persistent dynamic meetme To: asterisk-users at lists.digium.com Message-ID: <gumela$qs6$1 at ger.gmane.org> Content-Type: text/plain; charset=ISO-8859-1; format=flowed With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set a PIN. Hangup. Then users would dial 2666 enter the conference room number and the PIN, and be placed in conference. The problem is that as soon as 2663 hangs up, the conference disappears. What I'd like is to have the conference stay up for some period of time after 2663 hangs up. That way I could schedule a conference for a specific time, set it up beforehand, and email everyone the conference number and pin. I don't want the conference to stay up forever, since I'd like new pin's each time. This should be a common use case. How do you do it? sean ------------------------------ Message: 7 Date: Sat, 16 May 2009 14:49:36 +0100 From: David Anthony O Reilly <oreillda at tcd.ie> Subject: [asterisk-users] Agent-Login/out in 1.6 To: cursor at telecomabmex.com Cc: asterisk-users at lists.digium.com Message-ID: <f81bfd510905160649v281ad769pc2a65785f2553cc0 at mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Carlos " Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from 1.6 so you now need to use Dynamic Agents. Although they claim that is is simple enough to replace that functionality with dial plan code I have yet to see a one line example that replaces everything the agentcallbacklogin command did.| I totally agree, I have never seen any example that makes it work. If somebody shows me how to do it without using Voicemail I will let you know. Thanks David -- _________________________________________ Mr. David Anthony O'Reilly, B.Sc Comp (Hons) M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009 Computer Science Graduate of The University of Dublin, Trinity College - B.Sc (Comp) 2008 Email: oreillda at tcd.ie / dor3 at student.cs.ucc.ie Tel: +353 (0) 86 030 60 32 _________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/38 f1b815/attachment-0001.htm ------------------------------ Message: 8 Date: Sat, 16 May 2009 14:54:00 +0100 From: David Anthony O Reilly <oreillda at tcd.ie> Subject: [asterisk-users] Agent-Login/out in 1.6 To: dickenson at cfmc.com Cc: asterisk-users at lists.digium.com Message-ID: <f81bfd510905160654s421ccc7am1143414b0d6009c7 at mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Jim Thanks for your code!! I see you use the Voicemail system to authenticate, have you ever managed to avoid that as I don't use voicemail at all and I am thinking if I use that solution I will need to set up a voicemail for all the queue members just to get them to log in. hehe What were the developers thinking by removing the old system! It worked perfect!! and by the looks of it nobody has ever recovered from the command removal unless they hack around with the voicemail system. Hopefully somebody out there has managed to create an agent login/logout without bringing voicemail into it???? If I find a way I will let you and post a wiki on it as I am sure loads of people have this problem. Thanks Dave ; #### Agent login logout #### exten => *20,1,Answer() exten => *20,n,wait(.0.5) exten => *20,n,Read(AgentNumber,agent-user) exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})}) exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER) exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)}) exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY) exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY) exten => *20,n(NOUSER),Playback(cfmc/bad-agent) exten => *20,n,Playback(vm-goodbye) exten => *20,n,Hangup() exten => *20,n(VERIFY),VMAuthenticate(${AgentNumber}@ourvm) exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF) exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2) exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)}) exten => *20,n,AddQueueMember(support,Local/Queue${AgentNumber}@ansqueue,,,,${CUT (CHA NNEL,-,1)}) ; AQMSTATUS can be ADDED | MEMBERALREADY | NOSUCHQUEUE exten => *20,n,Playback(agent-loginok) exten => *20,n,Verbose(2,Agent ${AgentNumber} added ${DB(users/${UserID}/AgentDevice)}) exten => *20,n,HangUp() exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1) exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)}) exten => *20,n,RemoveQueueMember(support,Local/Queue${AgentNumber}@ansqueue) exten => *20,n,Playback(agent-loggedoff) exten => *20,n,Verbose(2,Agent ${AgentNumber} removed) exten => *20,n,Hangup()" -- _________________________________________ Mr. David Anthony O'Reilly, B.Sc Comp (Hons) M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009 Computer Science Graduate of The University of Dublin, Trinity College - B.Sc (Comp) 2008 Email: oreillda at tcd.ie / dor3 at student.cs.ucc.ie Tel: +353 (0) 86 030 60 32 _________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/5a 28fe3d/attachment-0001.htm ------------------------------ Message: 9 Date: Sat, 16 May 2009 16:06:04 +0200 From: Stefan Reuter <stefan.reuter at reucon.com> Subject: Re: [asterisk-users] Agent-Login/out in 1.6 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <4A0EC84C.3050405 at reucon.com> Content-Type: text/plain; charset="utf-8" David Anthony O Reilly wrote:> hehe What were the developers thinking by removing the old system! It > worked perfect!! and by the looks of it nobody has ever recovered from > the command removal unless they hack around with the voicemail system.I think the best solution is to either use an AGI script if agents should be able to login/logout through the phone, this allows you to store the agents in a database or any other external system. On the other hand many call centers use special applications for their agents and allow them to manage their availability through a GUI. Such applications can make use of the Manager API. They probably more convenient as they also allow to display additional information about the calls they handle and the queues they are subscribed to. =Stefan -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 260 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/ac 26f100/attachment-0001.pgp ------------------------------ Message: 10 Date: Sat, 16 May 2009 17:22:22 +0300 From: Timothy Smith <timotsmith at gmail.com> Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <416fc8170905160722n1e5d997dtcc244d827258fc3 at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" David, Thanks a lot for your input. I will enable DSP farming. Like some other techies, I just wanted to see it work before i consider others things. I have finally managed to get voice working. I both parties can hear each other. The problem was nating. Our network is fairly big and these machines are atleast 2 switches from each other. I just enabled it (nat=route or nat=yes) and it worked. It's not yet done however. When I redirect a call to any Asterisk application, it just hangs up! I have read some history and archives, but none of the solutions has worked for me. e.g ip inspect udp idle-time 900. My router (or IOS) doesn't have thet command. Could you please assist point to what could be causing this and how to solve it? Below are some logs and attached is the router log. ; This is the extension conf. Enter the extension you want to reach now (something like auto attendant). exten => _X.,1,Read(NUM,beep,4,2,3) exten => _X.,n,Dial(SIP/${NUM}) ; This is all i get when i call and the call hangs up! cs-intranet*CLI> == Using SIP RTP CoS mark 5 -- Executing [730732888 at default:1] Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack -- Accepting a maximum of 4 digits. == Using SIP RTP CoS mark 5 -- Executing [730732888 at default:1] Read("SIP/172.17.3.248-30069280", "NUM,beep,4,2,3") in new stack -- Accepting a maximum of 4 digits. cs-intranet*CLI> Thanks alot for your assistance. On Sat, May 16, 2009 at 4:02 PM, David Backeberg <dbackeberg at gmail.com> wrote:> On Sat, May 16, 2009 at 7:46 AM, Timothy Smith <timotsmith at gmail.com>wrote:>> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), >> and also a dialpeer to forward on the router to forward calls to my >> asterisk. It works properly but the problem is there is NO AUDIO! I >> have tried to change codec but no sucess! >> Has anyone had the above set up working successfully? > > Yes. > > You have been caught by a not-very-well-documented issue with setting > up voice routing on the 3845, and probably other similar Cisco gear. > And I'm not sure how you've done your test. > This is the closest I've ever seen to a document that explains yourproblem:>http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0 080147524.shtml> > Did you have a SIP phone on one side of asterisk and a POTS phone on > the outside of the 3845? > > If you did, and you could talk on both at the same time, I think you > would discover in fact that you do have some audio, in fact, one-way > audio to be precise. But I don't remember for sure, because it's been > a while since I've done this to myself. > > At any rate, your problem is you have dial-peers to get voice packets > out from the 3845 to Cisco, but no dial-peers to get the packets from > SIP back to a physical circuit on the 3845. Think about this. What > should happen to a call inbound from asterisk, to the 3845? Should it > go out an E1 to the outside phones world? If so, you need to build a > dial-peer that does that. Until you do, you won't be getting two-way > audio. > > you need another rule something like: > dial-peer voice 790792888 pots > map this back to the proper E1 circuit > > A secondary problem could also be with the way you're managing your > DSPs. I don't know how many physical DSPs you have in your router, but > usually it's a GOOD thing to enable DSP farming. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- May 16 14:18:28.640: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0CAB Cause i = 0x80D1 - Invalid call reference value May 16 14:18:28.640: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No NLCB 2 May 16 14:18:28.640: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8CAB callid 0x0 May 16 14:18:28.660: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref 0x0CAC Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9838D Exclusive, Channel 13 Calling Party Number i = 0x2183, '730730199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '730732888' Plan:ISDN, Type:National May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_INCOMING May 16 14:18:28.664: ISDN Se0/0/1:15 EVENT: call_incoming: call_id 0x3407, Guid = 42D2FC70B0D1 May 16 14:18:28.664: fb_get_reject_cause_code: ERROR cause_code NULL May 16 14:18:28.668: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 14:18:28.668: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8CAC Channel ID i = 0xA9838D Exclusive, Channel 13 May 16 14:18:28.676: ISDN Se0/0/0:15 SERROR: isdn_get_name_from_gtd: false ret May 16 14:18:28.676: ISDN Se0/0/1:15 SERROR: process_pri_simple: NO name in GTD May 16 14:18:28.676: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8CAC May 16 14:18:28.688: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0CAC May 16 14:18:28.688: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_PROGRESS May 16 14:18:28.688: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now connected to 730730199 N/A May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0CAC Cause i = 0x82AF - Resource unavailable, unspecified May 16 14:18:28.788: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_DISC May 16 14:18:28.788: %ISDN-6-CONNECT: Interface Serial0/0/1:12 is now connected to 730730199 N/A May 16 14:18:28.788: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8CAC May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000 Channel ID i = 0xA9838D Exclusive, Channel 13 Restart Indicator i = 0x80 May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_DISC May 16 14:18:28.792: ISDN Se0/0/1:15 SERROR: CCPRI_Go: call id 0x3407 event 0x57 No ccb Source->HOST May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x3407 calltype 2 CALL_CLEARED May 16 14:18:28.792: ISDN Se0/0/1:15 EVENT: process_rxstate: ces/callid 1/0x0 calltype 6 CHAN_STATUS OULKLAVG2# May 16 14:18:28.792: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000 Channel ID i = 0xA9838D Exclusive, Channel 13 Restart Indicator i = 0x80 May 16 14:18:28.800: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0CAC Cause i = 0x80D1 - Invalid call reference value May 16 14:18:28.800: ISDN Se0/0/1:15 SERROR: L3_GetUser_NLCB: EVENT 0X5A No NLCB 2 May 16 14:18:28.800: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8CAC callid 0x0 OULKLAVG2# ------------------------------ Message: 11 Date: Sat, 16 May 2009 07:27:32 -0700 From: Jim Dickenson <dickenson at cfmc.com> Subject: Re: [asterisk-users] Agent-Login/out in 1.6 To: David Anthony O Reilly <oreillda at tcd.ie> Cc: Asterisk User MailList <asterisk-users at lists.digium.com> Message-ID: <C6341B64.13C7D0%dickenson at cfmc.com> Content-Type: text/plain; charset="iso-8859-1" Where do you want to get the value that agent uses to validate? You can do your own code to do the validation. Get the value from where ever and then do a read and compare the value read with the value you retrieved from where ever. If there is match you are done if no match say error, maybe setup some counter and ask for value again. Or else use Authenticate(password[,options[,maxdigits]]): This application asks the caller to enter a given password in order to continue dialplan execution. If the password begins with the '/' character, it is interpreted as a file which contains a list of valid passwords, listed 1 password per line in the file. When using a database key, the value associated with the key can be anything. Users have three attempts to authenticate before the channel is hung up. Options: a - Set the channels' account code to the password that is entered d - Interpret the given path as database key, not a literal file m - Interpret the given path as a file which contains a list of account codes and password hashes delimited with ':', listed one per line in the file. When one of the passwords is matched, the channel will have its account code set to the corresponding account code in the file. r - Remove the database key upon successful entry (valid with 'd' only) maxdigits - maximum acceptable number of digits. Stops reading after maxdigits have been entered (without requiring the user to press the '#' key). Defaults to 0 - no limit - wait for the user press the '#' key. -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/ From: David Anthony O Reilly <oreillda at tcd.ie> Date: Sat, 16 May 2009 14:54:00 +0100 To: <dickenson at cfmc.com> Cc: <asterisk-users at lists.digium.com> Subject: Agent-Login/out in 1.6 Hi Jim Thanks for your code!! I see you use the Voicemail system to authenticate, have you ever managed to avoid that as I don't use voicemail at all and I am thinking if I use that solution I will need to set up a voicemail for all the queue members just to get them to log in. hehe What were the developers thinking by removing the old system! It worked perfect!! and by the looks of it nobody has ever recovered from the command removal unless they hack around with the voicemail system. Hopefully somebody out there has managed to create an agent login/logout without bringing voicemail into it???? If I find a way I will let you and post a wiki on it as I am sure loads of people have this problem. Thanks Dave ; #### Agent login logout #### exten => *20,1,Answer() exten => *20,n,wait(.0.5) exten => *20,n,Read(AgentNumber,agent- user) exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})}) exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER) exten => *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)}) exten => *20,n,GotoIf($["${AgentStatus}"="1"]?VERIFY) exten => *20,n,GotoIf($["${AgentStatus}"="2"]?VERIFY) exten => *20,n(NOUSER),Playback(cfmc/bad-agent) exten => *20,n,Playback(vm-goodbye) exten => *20,n,Hangup() exten => *20,n(VERIFY),VMAuthenticate(${AgentNumber}@ourvm) exten => *20,n,GotoIf($["${AgentStatus}"="2"]?AGENTOFF) exten => *20,n,Set(DB(users/${UserID}/AgentStatus)=2) exten => *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)}) exten => *20,n,AddQueueMember(support,Local/Queue${AgentNumber}@ansqueue,,,,${CUT (CHA NNEL,-,1)}) ; ? AQMSTATUS can be ?ADDED | MEMBERALREADY | NOSUCHQUEUE exten => *20,n,Playback(agent-loginok) exten => *20,n,Verbose(2,Agent ${AgentNumber} added ${DB(users/${UserID}/AgentDevice)}) exten => *20,n,HangUp() exten => *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1) exten => *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)}) exten => *20,n,RemoveQueueMember(support,Local/Queue${AgentNumber}@ansqueue) exten => *20,n,Playback(agent-loggedoff) exten => *20,n,Verbose(2,Agent ${AgentNumber} removed) exten => *20,n,Hangup()" -- _________________________________________ Mr. David Anthony O'Reilly, B.Sc Comp (Hons) M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) - 2009 Computer Science Graduate of The University of Dublin, Trinity College - B.Sc (Comp) 2008 Email: oreillda at tcd.ie ? ?/ ? ?dor3 at student.cs.ucc.ie Tel: +353 (0) 86 030 60 32 _________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/f5 0d20ac/attachment-0001.htm ------------------------------ Message: 12 Date: Sat, 16 May 2009 10:17:54 -0500 From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com> Subject: Re: [asterisk-users] howto set up persistent dynamic meetme To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <200905161017.54997.tilghman at mail.jeffandtilghman.com> Content-Type: text/plain; charset="iso-8859-1" On Saturday 16 May 2009 08:21:43 sean darcy wrote:> With 1.6.1, I'm trying to set up a test of meetme for creating dynamic > conferences. > > extensions.conf: > > [meetme] > exten => 2663,1,MeetMe(,De) > exten => 2663,n,Hangup() > > exten => 2666,1,MeetMe() > exten => 2666,n,Hangup() > > > What I'm expecting is to dial 2663, get a conference room number (600,> I suppose since it's the only room ), and set a PIN. Hangup. > > Then users would dial 2666 enter the conference room number and thePIN,> and be placed in conference. > > The problem is that as soon as 2663 hangs up, the conferencedisappears.> What I'd like is to have the conference stay up for some period oftime> after 2663 hangs up. That way I could schedule a conference for a > specific time, set it up beforehand, and email everyone the conference > number and pin. > > I don't want the conference to stay up forever, since I'd like newpin's> each time. > > This should be a common use case. How do you do it?In Asterisk 1.6, user DEA contributed realtime capabilities to MeetMe, which allows the capability of scheduling conferences, with new pins each time. I believe this would meet the needs your question has posed. -- Tilghman ------------------------------ Message: 13 Date: Sat, 16 May 2009 17:51:39 +0200 From: Philipp Kempgen <philipp.kempgen at amooma.de> Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <4A0EE10B.1080003 at amooma.de> Content-Type: text/plain; charset=ISO-8859-1 Steve Howes schrieb:> Check about the sip.conf 'insecure' option. I have had to use it in > the past for similar stuff. I think it was 'insecure=very' but that > might be deprecated by now..insecure=very should now be written as insecure=port,invite Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de -- ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 58, Issue 40 **********************************************
through a test .. i was able to send calls from Asterisk 1.4 to a PSTN number through a cisco router with a channel bank.. Audio worked well.. i setup a dial plan in asterisk to Dial(${EXTEN}@ciscoip) and authorise the cisco router's ip on the asterisk server and treat the calls comming from it like any other SIP calls inside the server.. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308> Date: Sat, 16 May 2009 14:46:27 +0300 > From: timotsmith at gmail.com > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router > > Hi, > > In our office, we're slowly migrating from a cisco call manager set up > to asterisk. Problem is management doesn't want to buy any other > hardware as they had already invested a lot in cisco. The main cause > of this is asterisk's added features like unique FAX number for > everyone in the company (which will be the same as phone DID), Voice > mail, Auto Answer etc yet we need thousands of dollars to add those to > our cisco call manager 4.1 set up. > > I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), > and also a dialpeer to forward on the router to forward calls to my > asterisk. It works properly but the problem is there is NO AUDIO! I > have tried to change codec but no sucess! > > Has anyone had the above set up working successfully? Attached are some confs. > > Thanks a lot for your assistance. > > Kind Regards, > Wilson_________________________________________________________________ Insert movie times and more without leaving Hotmail?. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutorial_QuickAdd1_052009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090526/b740b3dc/attachment.htm